Asterisk Development Team
2016-Mar-29 22:12 UTC
[asterisk-users] Asterisk 13.8.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 13.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
Journo)
* ASTERISK-25480 - [patch]Add field PauseReason on
QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25849 - chan_pjsip: transfers with direct media
sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard
system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in
update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache
(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
PJSIP requirement (Reported by Gergely D??ms??di)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
OutboundSubscriptionDetail ami action (Reported by Kevin
Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and
heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support
pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery
(Reported by Martin Mou??ka)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option
not in Alembic (Reported by Joshua Colp)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in
weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Reported by Mark
Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and
Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
test sporadically failing (Reported by Joshua Colp)
* ASTERISK-24097 - Documentation - CHANNEL function help text
missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer
fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by
Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25317 - asterisk sends too many stun requests (Reported
by Stefan Engstr??m)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-25495 - [patch] Prevent old-update packages on
repository Debian systems (Reported by Rodrigo Ramirez
Norambuena)
* ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
(Reported by Andrew Nagy)
* ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
Messina)
* ASTERISK-24813 - asterisk.c: #if statement in listener()
confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
core set (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0
Thank you for your continued support of Asterisk!
there is no info about --with-pjproject-bundled
i tried it (centos6 32bit)
./configure --with-pjproject-bundled
checking for SSL_library_init in -lssl... yes
OpenSSL library found, SSL support enabled
./aconfigure: line 14995: syntax error near unexpected token `fi'
./aconfigure: line 14995: `fi'
make: *** [build.mak] Error 2
make: Leaving directory
`/root/rpmbuild/SOURCES/asterisk-13.8.0/third-party/pjproject'
vim ./third-party/pjproject/source/aconfigure
# Check whether --with-opencore-amrnb was given.
if test "${with_opencore_amrnb+set}" = set; then
withval=$with_opencore_amrnb; { { $as_echo "$as_me:$LINENO: error:
This option is obsolete and replaced by --with-opencore-amr=DIR" >&5
$as_echo "$as_me: error: This option is obsolete and replaced by
--with-opencore-amr=DIR" >&2;}
{ (exit 1); exit 1; }; }
else
???missing something???
fi
Dne 30.3.2016 v 0:12 Asterisk Development Team
napsal(a):> The Asterisk Development Team has announced the release of Asterisk 13.8.0.
> This release is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 13.8.0 resolves several issues reported by the
> community and would have not been possible without your participation.
> Thank you!
>
> The following are the issues resolved in this release:
>
> New Features made in this release:
> -----------------------------------
> * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
> contents to file (Reported by Ray Crumrine)
> * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
> Journo)
> * ASTERISK-25480 - [patch]Add field PauseReason on
> QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
>
> Bugs fixed in this release:
> -----------------------------------
> * ASTERISK-25849 - chan_pjsip: transfers with direct media
> sometimes drops audio (Reported by Kevin Harwell)
> * ASTERISK-25113 - install_prereq in Debian 8 without "standard
> system utilities" (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
> (Reported by Sergio Medina Toledo)
> * ASTERISK-25023 - Deadlock in chan_sip in
> update_provisional_keepalive (Reported by Arnd Schmitter)
> * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
> channel (Reported by Filip Frank)
> * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
> separating multiple AORs (Reported by Mateusz Kowalski)
> * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
> Stasis application. (Reported by Javier Riveros )
> * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
> Bright)
> * ASTERISK-25582 - Testsuite: Reactor timeout error in
> tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
> Jordan)
> * ASTERISK-25811 - Unable to delete object from sorcery cache
> (Reported by Ross Beer)
> * ASTERISK-25800 - [patch] Calculate talktime when is first call
> answered (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
> PJSIP requirement (Reported by Gergely D??ms??di)
> * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
> when calling from Gosub (Reported by Jacques Peacock)
> * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
> OutboundSubscriptionDetail ami action (Reported by Kevin
> Harwell)
> * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
> heap-use-after-free (Reported by Badalian Vyacheslav)
> * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
> returns garbage (Reported by Etienne Lessard)
> * ASTERISK-25751 - res_pjsip: Support
> pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
> * ASTERISK-25606 - Core dump when using transports in sorcery
> (Reported by Martin Mou??ka)
> * ASTERISK-20987 - non-admin users, who join muted conference are
> not being muted (Reported by hristo)
> * ASTERISK-25737 - res_pjsip_outbound_registration: line option
> not in Alembic (Reported by Joshua Colp)
> * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
> udptl_rx_packet cause ast_frdup crash (Reported by Walter
> Doekes)
> * ASTERISK-25742 - Secondary IFP Packets can result in accessing
> uninitialized pointers and a crash (Reported by Torrey Searle)
> * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
> Vulnerability - Investigate vulnerability of HTTP server
> (Reported by Alex A. Welzl)
> * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
> non-default timert1 (Reported by Alexander Traud)
> * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
> upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
> Nic Colledge)
> * ASTERISK-25730 - build: make uninstall after make distclean
> tries to remove root (Reported by George Joseph)
> * ASTERISK-25725 - core: Incorrect XML documentation may result in
> weird behavior (Reported by Joshua Colp)
> * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
> sip_sipredirect (Reported by Badalian Vyacheslav)
> * ASTERISK-25709 - ARI: Crash can occur due to race condition when
> attempting to operate on a hung up channel (Reported by Mark
> Michelson)
> * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
> by Badalian Vyacheslav)
> * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
> script (Reported by Joshua Colp)
> * ASTERISK-25712 - Second call to already-on-call phone and
> Asterisk sends "Ready" (Reported by Richard Mudgett)
> * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
> incorrect values (Reported by Gianluca Merlo)
> * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
> test sporadically failing (Reported by Joshua Colp)
> * ASTERISK-24097 - Documentation - CHANNEL function help text
> missing 'linkedid' argument (Reported by Steven T. Wheeler)
> * ASTERISK-25700 - main/config: Clean config maps on shutdown.
> (Reported by Corey Farrell)
> * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
> a transfer (Reported by Kevin Harwell)
> * ASTERISK-25697 - bridge_basic: don't play an attended transfer
> fail sound after target hangs up (Reported by Kevin Harwell)
> * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
> with MALLOC_DEBUG (Reported by yaron nahum)
> * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
> schema is an integer (Reported by Marcelo Terres)
> * ASTERISK-25690 - Hanging up when executing connected line sub
> does not cause hangup (Reported by Joshua Colp)
> * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
> reload' cause a crash (Reported by Sean Bright)
> * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
> address when multihomed (Reported by Olivier Krief)
> * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
> Daniel Journo)
> * ASTERISK-25394 - pbx: Incorrect device and presence state when
> changing hint details (Reported by Joshua Colp)
> * ASTERISK-25640 - pbx: Deadlock on features reload and state
> change hint. (Reported by Krzysztof Trempala)
> * ASTERISK-25681 - devicestate: Engine thread is not shut down
> (Reported by Corey Farrell)
> * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
> shutdown (Reported by Corey Farrell)
> * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
> Corey Farrell)
> * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
> Daniel Journo)
> * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
> by Corey Farrell)
> * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
> Farrell)
> * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
> Mark Michelson)
> * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
> (Reported by Corey Farrell)
> * ASTERISK-25647 - bug of cel_radius.c: wrong point of
> ADD_VENDOR_CODE (Reported by Aaron An)
> * ASTERISK-25317 - asterisk sends too many stun requests (Reported
> by Stefan Engstr??m)
> * ASTERISK-25137 - endpoint stasis messages are delivered twice
> (Reported by Vitezslav Novy)
> * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
> sent for every status change (Reported by George Joseph)
> * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
> transfer initiated channel (Reported by Dmitry Melekhov)
> * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
> Brandon)
> * ASTERISK-25442 - using realtime (mysql) queue members are never
> updated in wait_our_turn function (app_queue.c) (Reported by
> Carlos Oliva)
> * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
> caching (Reported by Joshua Colp)
> * ASTERISK-25601 - json: Audit reference usage and thread safety
> (Reported by Joshua Colp)
> * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
> sungtae kim)
>
> Improvements made in this release:
> -----------------------------------
> * ASTERISK-25495 - [patch] Prevent old-update packages on
> repository Debian systems (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
> (Reported by Andrew Nagy)
> * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
> Anonymous <anonymous at anonymous.invalid> (Reported by
Anthony
> Messina)
> * ASTERISK-24813 - asterisk.c: #if statement in listener()
> confuses code folding editors (Reported by Corey Farrell)
> * ASTERISK-25767 - [patch] Add check to configure for sanitizes
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
> core set (Reported by Rusty Newton)
>
> For a full list of changes in this release, please see the ChangeLog:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0
>
> Thank you for your continued support of Asterisk!
>
>
>
--
---------------------------------------
Marek Cervenka
======================================
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and what about https://www.asterisk-blog.com/2016/02/17/odbc_gutting/ Dne 30.3.2016 v 0:12 Asterisk Development Team napsal(a):> The Asterisk Development Team has announced the release of Asterisk 13.8.0. > This release is available for immediate download at > http://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 13.8.0 resolves several issues reported by the > community and would have not been possible without your participation. > Thank you! > > The following are the issues resolved in this release: > > New Features made in this release: > ----------------------------------- > * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write > contents to file (Reported by Ray Crumrine) > * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel > Journo) > * ASTERISK-25480 - [patch]Add field PauseReason on > QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) > > Bugs fixed in this release: > ----------------------------------- > * ASTERISK-25849 - chan_pjsip: transfers with direct media > sometimes drops audio (Reported by Kevin Harwell) > * ASTERISK-25113 - install_prereq in Debian 8 without "standard > system utilities" (Reported by Rodrigo Ramirez Norambuena) > * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so > (Reported by Sergio Medina Toledo) > * ASTERISK-25023 - Deadlock in chan_sip in > update_provisional_keepalive (Reported by Arnd Schmitter) > * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local > channel (Reported by Filip Frank) > * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when > separating multiple AORs (Reported by Mateusz Kowalski) > * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into > Stasis application. (Reported by Javier Riveros ) > * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean > Bright) > * ASTERISK-25582 - Testsuite: Reactor timeout error in > tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt > Jordan) > * ASTERISK-25811 - Unable to delete object from sorcery cache > (Reported by Ross Beer) > * ASTERISK-25800 - [patch] Calculate talktime when is first call > answered (Reported by Rodrigo Ramirez Norambuena) > * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to > PJSIP requirement (Reported by Gergely D??ms??di) > * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity > when calling from Gosub (Reported by Jacques Peacock) > * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing > OutboundSubscriptionDetail ami action (Reported by Kevin > Harwell) > * ASTERISK-25721 - [patch] res_phoneprov: memory leak and > heap-use-after-free (Reported by Badalian Vyacheslav) > * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes > returns garbage (Reported by Etienne Lessard) > * ASTERISK-25751 - res_pjsip: Support > pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp) > * ASTERISK-25606 - Core dump when using transports in sorcery > (Reported by Martin Mou??ka) > * ASTERISK-20987 - non-admin users, who join muted conference are > not being muted (Reported by hristo) > * ASTERISK-25737 - res_pjsip_outbound_registration: line option > not in Alembic (Reported by Joshua Colp) > * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in > udptl_rx_packet cause ast_frdup crash (Reported by Walter > Doekes) > * ASTERISK-25742 - Secondary IFP Packets can result in accessing > uninitialized pointers and a crash (Reported by Torrey Searle) > * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST > Vulnerability - Investigate vulnerability of HTTP server > (Reported by Alex A. Welzl) > * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with > non-default timert1 (Reported by Alexander Traud) > * ASTERISK-25702 - PjSip realtime DB and Cache Errors since > upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by > Nic Colledge) > * ASTERISK-25730 - build: make uninstall after make distclean > tries to remove root (Reported by George Joseph) > * ASTERISK-25725 - core: Incorrect XML documentation may result in > weird behavior (Reported by Joshua Colp) > * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in > sip_sipredirect (Reported by Badalian Vyacheslav) > * ASTERISK-25709 - ARI: Crash can occur due to race condition when > attempting to operate on a hung up channel (Reported by Mark > Michelson) > * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported > by Badalian Vyacheslav) > * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build > script (Reported by Joshua Colp) > * ASTERISK-25712 - Second call to already-on-call phone and > Asterisk sends "Ready" (Reported by Richard Mudgett) > * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow > (Reported by Badalian Vyacheslav) > * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report > incorrect values (Reported by Gianluca Merlo) > * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit > test sporadically failing (Reported by Joshua Colp) > * ASTERISK-24097 - Documentation - CHANNEL function help text > missing 'linkedid' argument (Reported by Steven T. Wheeler) > * ASTERISK-25700 - main/config: Clean config maps on shutdown. > (Reported by Corey Farrell) > * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during > a transfer (Reported by Kevin Harwell) > * ASTERISK-25697 - bridge_basic: don't play an attended transfer > fail sound after target hangs up (Reported by Kevin Harwell) > * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled > with MALLOC_DEBUG (Reported by yaron nahum) > * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database > schema is an integer (Reported by Marcelo Terres) > * ASTERISK-25690 - Hanging up when executing connected line sub > does not cause hangup (Reported by Joshua Colp) > * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh > reload' cause a crash (Reported by Sean Bright) > * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP > address when multihomed (Reported by Olivier Krief) > * ASTERISK-25637 - Multi homed server using wrong IP (Reported by > Daniel Journo) > * ASTERISK-25394 - pbx: Incorrect device and presence state when > changing hint details (Reported by Joshua Colp) > * ASTERISK-25640 - pbx: Deadlock on features reload and state > change hint. (Reported by Krzysztof Trempala) > * ASTERISK-25681 - devicestate: Engine thread is not shut down > (Reported by Corey Farrell) > * ASTERISK-25680 - manager: manager_channelvars is not cleaned at > shutdown (Reported by Corey Farrell) > * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by > Corey Farrell) > * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by > Daniel Journo) > * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported > by Corey Farrell) > * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey > Farrell) > * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by > Mark Michelson) > * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference > (Reported by Corey Farrell) > * ASTERISK-25647 - bug of cel_radius.c: wrong point of > ADD_VENDOR_CODE (Reported by Aaron An) > * ASTERISK-25317 - asterisk sends too many stun requests (Reported > by Stefan Engstr??m) > * ASTERISK-25137 - endpoint stasis messages are delivered twice > (Reported by Vitezslav Novy) > * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are > sent for every status change (Reported by George Joseph) > * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on > transfer initiated channel (Reported by Dmitry Melekhov) > * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade > Brandon) > * ASTERISK-25442 - using realtime (mysql) queue members are never > updated in wait_our_turn function (app_queue.c) (Reported by > Carlos Oliva) > * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend > caching (Reported by Joshua Colp) > * ASTERISK-25601 - json: Audit reference usage and thread safety > (Reported by Joshua Colp) > * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by > sungtae kim) > > Improvements made in this release: > ----------------------------------- > * ASTERISK-25495 - [patch] Prevent old-update packages on > repository Debian systems (Reported by Rodrigo Ramirez > Norambuena) > * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps > (Reported by Andrew Nagy) > * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for > Anonymous <anonymous at anonymous.invalid> (Reported by Anthony > Messina) > * ASTERISK-24813 - asterisk.c: #if statement in listener() > confuses code folding editors (Reported by Corey Farrell) > * ASTERISK-25767 - [patch] Add check to configure for sanitizes > (Reported by Badalian Vyacheslav) > * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the > core set (Reported by Rusty Newton) > > For a full list of changes in this release, please see the ChangeLog: > > http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0 > > Thank you for your continued support of Asterisk! > > >-- --------------------------------------- Marek Cervenka ====================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160330/6fa85dd4/attachment.html>
Marek ?ervenka wrote:> and what about > https://www.asterisk-blog.com/2016/02/17/odbc_gutting/While not in the email these are listed in the CHANGES and UPGRADE.txt file. Going forward we'll try to ensure we include such things in the release notes as well. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
Have you ever heard of Asterisk Development.There are only few companies in India which are providing this service and "Anticlock Technologies is one of them.it is dealing in this field from long time and We provide client satisfaction, full support and long term services .
Jeff LaCoursiere
2016-Mar-31 16:01 UTC
[asterisk-users] Asterisk Development Company in India
And punctuation and grammar skills have we too! Our english be VERY good On 03/31/2016 02:20 AM, ankur verma wrote:> Have you ever heard of Asterisk Development.There are only few companies in > India which are providing this service and "Anticlock Technologies is one of > them.it is dealing in this field from long time and We provide client > satisfaction, full support and long term services > . > >